Audio synthesizing systems and methods

ABSTRACT

A system and method is disclosed teach how to synthesizing audio. It allows specification of a musical sound to be generated. It synthesizes an audio source, such as noise, using parameters to specify the desired frequency slit spacing and the desired noise-to-frequency band ratio, then filtering the audio source through a sequence of filters to obtain the desired frequency slit spacing and noise to frequency band ratio. It allows modulation of the filters in the sequence. It outputs musical sound.

CROSS REFERENCE TO RELATED APPLICATIONS

The present application is related to utility application entitled“MUSIC SOFTWARE SYSTEMS”, filed Aug. 2, 2010, bearing attorney docketnumber 10#357, bearing Ser. No. 61/400,817, the contents of which areincorporated herein by this reference and are not admitted to be priorart with respect to the present invention by the mention in thiscross-reference section.

FIELD OF INVENTION

Embodiments of the invention are generally related to music, audio, andother sound processing and synthesis, and are particularly related to asystem and method for audio synthesis.

SUMMARY

Disclosed herein is a system and method for audio synthesizer utilizingfrequency aperture cells (FAC) and frequency aperture arrays (FAA). Inaccordance with an embodiment, an audio processing system can beprovided for the transformation of audio-band frequencies for musicaland other purposes. In accordance with an embodiment, a single stream ofmono, stereo, or multi-channel monophonic audio can be transformed intopolyphonic music, based on a desired target musical note or set ofmultiple notes. The system utilizes an input waveform(s) (which can beeither file-based or streamed) which is then fed into an array offilters, which are themselves optionally modulated, to generate a newsynthesized audio output.

Previous techniques for dealing with both pitched and non-pitched audioinput is known as subtractive synthesis, whereby single or multi-poleHigh Pass, Low Pass, Band Pass, Resonant and non-resonant filters areused to subtract certain unwanted portions from the incoming sound. Inthis technique, the subtractive filters usually modify the perceivedtimbre of the note, however the filter process does not determine theperceived pitch, except in the unusual ease of extreme filter resonance.These filters are usually of type IIR, Infinite Impulse Response,indicating a delay line and a feedback path. Others who have employednoise routed through IIR filters are Kevin Karplus, Alex Strong (1983).“Digital Synthesis of Plucked String and Drum Timbres”. Computer Musicjournal (MIT Press) 7 (2): 43-55. doi:10.2307/3680062, incorporatedherein by reference. Although arguably also subtractive, in theseprevious techniques the resonance of the filter usually determines thepitch as well as it affects the timbre. There have been variousimprovements to these previous techniques, whereby certain filterdesigns are intended to emulate certain portions of their acousticcounterparts.

Compared to additive synthesis, the present invention allow for greatercomputational efficiency and facilitation of the synthesis of noisesound components as they combine and modulate in complex ways. Bysynthesizing groups of harmonic and inharmonic related frequencies,rather than individually synthesizing each individual frequency partial,significant computational efficiencies can be gained, and more costeffective systems can be built. Additive synthesis does not have theability to produce realistic noise components nor has it the ability forcomplex noise interactions, as is desirable for many types of musicalsounds.

Advantages of various embodiments of the present invention over previoustechniques include that the input audio source can be completelyunpitched and unmusical, even consisting of just pure white noise or aperson's whisper, and after being synthesized by the FAA have theability to be completely musical, with easily recognized pitch andtimbre components; and the use of a real-time streamed audio input togenerate the input source which is to be synthesized. The frequencyaperture synthesis approach allows for both file-based audio sources andreal-time streamed input. The result is a completely new sound withunlimited scope because the input source itself has unlimited scope. Inaccordance with an embodiment, the system also allows multiple synthesisto be combined to create unique hybrid sounds, or accept input from amusical keyboard, as an additional input source to the FAA filters.Other features and advantages will be evident from the followingdescription.

BRIEF DESCRIPTION OF THE DRAWINGS:

FIG. 1 illustrates a block-diagram view showing a 3-series-by-2-parallelarray of frequency aperture cells (FACs), in accordance with anembodiment.

FIG. 2 illustrates a block-diagram view showing ann-series-by-m-parallel array of frequency aperture cells (FACs), inaccordance with an embodiment.

FIG. 3 illustrates a block-diagram view an isolated frequency aperturecell (FAC) within an frequency aperture array, along with deviceconnections, in accordance with an embodiment.

FIG. 4 a illustrates a block-diagram view showing an example of afrequency aperture filter in accordance with an embodiment.

FIG. 4 b illustrates a block-diagram view showing another example of afrequency aperture filter in accordance with another embodiment.

FIG. 5 illustrates a block-diagram view showing the selection andcombination block of FIGS. 4 a and 4 b in accordance with an embodiment.

FIG. 6 illustrates a block-diagram view showing the interpolate andprocess block of FIGS. 4 a and 4 b in accordance with an embodiment.

FIG. 7 illustrates a block-diagram view showing one example of amulti-mode filter, which may be used in FIGS. 4 a and 4 b in accordancewith an embodiment.

FIG. 8 illustrates a block-diagram view showing various modulators inaccordance with an embodiment.

FIG. 9 illustrates a block-diagram view showing the stabilitycompensation filter of FIG. 5 in accordance with an embodiment.

FIG. 10 illustrates a block-diagram view showing how an audio inputsource into the FAA synthesizer can be modulated before entering the FAAfilters, and how the FAA filters themselves can be modulated inreal-time, in accordance with an embodiment.

FIG. 11 a illustrates a FFT spectral waveform graph view showing aslit_height of 100% in accordance with an embodiment.

FIG. 11 b illustrates a FFT spectral waveform graph view showing aslit_height of 50% in accordance with an embodiment.

FIG. 11 c illustrates a FFT spectral waveform graph view showing aslit_height of 0% in accordance with an embodiment.

FIG. 11 d illustrates a FFT spectral waveform graph view showing aslit_height of −50% in accordance with an embodiment.

FIG. 11 e illustrates a FFT spectral waveform graph view showing aslit_height of −100% in accordance with an embodiment.

FIG. 12 illustrates a FFT spectral waveform graph view showing acomparison of brown noise and pink noise as audio input in accordancewith an embodiment.

FIG. 13 illustrates a FFT spectral waveform graph view showing a seriesof waveforms in a 1-series-by-2-parallel array, including waveforms foraudio input, waveforms for output from each FAC, and a final waveformfor audio output in accordance with an embodiment.

FIG. 14 illustrates a FFT spectral waveform graph view showing a seriesof waveforms in a 2-series-by-1-parallel array, including a waveform foraudio input, waveforms for output from each FAC, and a final waveformfor audio output in accordance with an embodiment.

FIG. 15 illustrates a FFT spectral waveform graph view showing a seriesof waveforms in a 1-series-by-2-parallel array, including identicalwaveforms for audio input, waveforms for output from each FAC, eachprocessed separately with a different FAF Type, and each showingdifferent final waveforms for audio output in accordance with anembodiment.

FIGS. 1, 17, 18, 19, and 20 illustrate a series of computer screenshotviews showing user controls to select parameters, such as slit_height,slit_width and other pre-sets, for use or initialization in the FACs inaccordance with an embodiment.

Appendix A lists sets of parameters and other pre-sets to producevarious example timbres in accordance with an embodiment.

DETAILED DESCRIPTION

Disclosed herein is a system and method for audio synthesizer utilizingfrequency aperture cells (FAC) and frequency aperture arrays (FAA). Inaccordance with an embodiment, an audio processing system can beprovided for the transformation of audio-band frequencies for musicaland other purposes. In accordance with an embodiment, a single stream ofmono, stereo, or multi--channel monophonic audio can be transformed intopolyphonic music, based on a desired target musical note or set ofmultiple notes. At its core, the system utilizes an input waveform(s)(which can be either file-based or streamed) which is then fed into anarray of filters, which are themselves optionally modulated, to generatea new synthesized audio output.

FIG. 1 illustrates a block-diagram view showing a 3-series-by-2-parallelarray of frequency aperture cells (FACs) 110, in accordance with anembodiment; while FIG. 2 illustrates a block-diagram view showing ann-series-by-m-parallel array of frequency aperture cells (FACs) 110, inaccordance with an embodiment. These figures show how filtering theaudio source through a sequence of filters creates a series offrequency-bands-with-noise, where the first filter receives the audiosource and subsequent filters receive the output of the previous filteras input, with the last filter producing audio output for the system. Asshown in FIGS. 1 and 2, each array is organized into n rows by mcolumns, representing n successive series connections of audioprocessing, the output of which is then summed with m parallel rows ofprocessing. A channel of mono, stereo, or multi-channel source audio 130feeds each row. The source audio 130 may be live audio or pre-loadedfrom a file storage system, such as on the hard drive of a personalcomputer.

In accordance with an embodiment, frequency aperture arrays 100 (FAAs)may be organized into n series by m parallel connections of frequencyaperture cells, and optionally other digital filters such as multimodehigh pass (HP), band pass (BP), low pass (LP), or band restrict (BR)filters, or resonators of varying type, or combinations. In otherembodiments, the multi-mode filter may be omitted.

An advantage of various embodiments of the present invention overprevious techniques is how the input audio source 130 can be completelyunpitched or unmusical, for example, pure white noise or a person'swhisper, and after being synthesized have the ability to be musical,with recognized pitch and timbre components. The output audio 140 isunlimited in its scope, and can include realistic instrument sounds suchas violins, piano, brass instruments, etc., electronic sounds, soundeffects, and sounds never conceived or heard before.

Previously, musical synthesizers have relied upon stored files (usuallypitched) which consist of audio waveforms, either recorded (sample basedsynthesis) or algorithmically generated (frequency or amplitudemodulated synthesis) to provide the audio source which is thensynthesized.

By comparison, the systems and methods disclosed herein allow the audioinput 130 to be file-based audio sources, real-time streamed input, orcombinations. The resulting audio output 140 can be a completely newsound with unlimited scope, in part, because the input source 130 hasunlimited scope.

In accordance with an embodiment, the system provides advantages overprior musical synthesis, by employing arrays 100 of frequency aperturecells 110 (FAC) which contain frequency aperture filters (FAF) (SeeFIGS. 4 a, 4 b and accompanying text). FACs 110 have the ability totransform a spectrum of related or unrelated, harmonic or inharmonicinput frequencies into an arbitrary, and potentially continuouslychanging set of new output frequencies. There are no constraints on thetype of filter designs employed, only that they have inherent slits ofharmonic or in-harmonic frequency bands that separate desired frequencycomponents between their input and output. Both FIR (Finite ImpulseResponse) and IIR (Infinite Impulse Response) type filter designs areemployed within different embodiments of the FAC 110 types. In otherembodiments, additive or subtractive filters may be employed. Musicallyinteresting effects are obtained as individual frequency slit width,analogous to frequency spacing, and height, analogous to amplitude, arevaried between FAC no stages. Frequency slit spacing is a collection ofharmonic and/or inharmonic frequency components, for example, harmonicpartial frequency would be an example of substantially harmonic. FAC 110stages are connected in series and in parallel, and can each bemodulated by specific modulation signals, such as LFO's, Envelopegenerators, or by the outputs of prior stages. (See FIGS. 4 a, 4 b, 8and accompanying text.) This demonstrates how to modulate the output ofa frequency aperture filter in the sequence using a modulator such aslow frequency oscillator modulator, random generator modulator, envelopmodulator, and MIDI control modulator,

Frequency spacing from the output of the FAC 110 is often not even (i.e.harmonic), hence the term “slit width” instead of “pitch” is used. “Slitwidth” can affect both the pitch, timbre or just one or the other, sothe use of “pitch” is not appropriate in the context of an FAC 110array.

In some embodiments, each frequency aperture cell 110 in the array iscomprised of its own set of modulators having separate parameters slitwidth, slit height and amplitude, as well as audio input, a cascadeinput, an audio output, transient impulse scaling, and a FrequencyAperture Filter (FAF) (See FIGS. 4 a, 4 b and accompanying text).

Other advantages of embodiments of the present invention over previoustechniques is the use of a real-time streamed audio input to generatethe input source 130 which is to be synthesized. In order to facilitatepitched streamed audio input sources 130, in accordance with anembodiment, the system also includes a dispersion algorithm which cantake a pitched input source and make it unpitched and noise-like (broadspectrum). This signal then feeds into the system which furthersynthesizes the audio signal. This allows for a unique attribute inwhich a person can sing, whisper, talk or vocalize into the dispersionfilter, which, when fed into the system and triggered by a keyboard orother source guiding the pitch components of the system synthesizer, canyield an output that sounds like anything, including a real instrumentsuch as a piano, guitar, drumset, etc. The input source 130 is notlimited to vocalizations of course. Any pitched input source (guitar,drumset, piano, etc.) can be dispersed into broad spectrum noise andre-synthesized to produce any musical instrument output, for example,using a guitar as input, dispersing the guitar into noise, andre-synthesizing into a piano. This demonstrates how the system can usenon-pitched, broad-spectrum audio with no discernible pitch and timbre;and the audio output becomes pitched, musical sounds with discerniblepitch and timbre.

The input audio signal 130 can consist of any audio source in any formatand be read in via a file-based system or streamed audio. A file-basedinput may include just the raw PCM data or the PCM data along withinitial states of the FAA filter parameters and/or modulation data.

In accordance with an embodiment, the system also allows multiplesynthesis to be combined to create unique hybrid sounds. Finally,embodiments of the invention include a method of using multiple impulseresponses, mapped out across a musical keyboard, as an additional inputsource to the FAA filters, designed, but not limited to, synthesizingthe first moments of a sound.

FIG. 3 illustrates a block-diagram view showing an isolated frequencyaperture cell 210 (FAC) within an frequency aperture array, along withdevice connections, in accordance with an embodiment. In accordance withan embodiment, the system uses an array of audio frequency aperturecells 200, which separate noise components into harmonic and inharmonicfrequency multiples. Storage of control parameters 210, such asmodulation and other musical controls, and source or impulse transientaudio files come from a storage system 220, such as a hard drive orother storage device. A unique set of each of these files and parametersis loaded into runtime memory for each Frequency Aperture Cell 210 inthe array. The system may be built of software, hardware, or acombination of both. With the data packed and unpacked into interleavechannels of data (e.g. RAM Stereo Circular Buffer 230), four channelscan be processed simultaneously.

Each frequency aperture cell 200, with varying feedback properties,produces instantaneous output frequency based on both the instantaneousspectrum of incoming audio, as well as the specific frequency slits andresonance of the aperture filter. Two controlling properties are thefrequency slit spacing (slit width) 240 and the noise-to-frequency bandratio, or frequency (slit height) 250.

An important distinction of constituent FAA cells 200 is that their slitwidths 240 are not necessarily representative of the pitch of theperceived audio output. FAA cells 200 may be inharmonic themselves, orin the case of two or more series cascaded harmonic cells of differingslit width 240, they may have their aperture slits at non-harmonicrelationships, producing inharmonic transformations through cascadedharmonic cells. The perceived pitch is often a complex relationship ofthe slit widths and heights of all constituent cells and the characterof their individual harmonic and inharmonic apertures. The slit width240 and height 250 are as important to the timbre of the audio as theyare to the resultant pitch.

In accordance with an embodiment, this system and method are provided byemploying arrays of frequency aperture cells 200. FACs 200 have theability to transform a spectrum of related or unrelated, harmonic orinharmonic input frequencies into an arbitrary, and potentiallycontinuously changing set of new output frequencies. There are noconstraints on the type of filter designs employed, only that they haveinherent slits of harmonic or in-harmonic frequency bands that separatedesired frequency components between their input and output. Both FIR(Finite Impulse Response) and IIR (Infinite Impulse Response) typedesigns are employed within different embodiments of the FAA types.Musically interesting effects are obtained as individual frequency slitwidth, analogous to frequency spacing, and height, analogous toamplitude, are varied between FAC 200 stages. This demonstrates howvarying the parameters between the filters in the sequence is useful.

In accordance with an embodiment, FAC 200 stages are connected in seriesand in parallel, and can each be modulated by specific modulationsignals, such as LFO's, Envelope generators, or by the outputs of priorstages. This demonstrates how to modulate the output of a filters in thesequence using the output of another filter in the sequence, forexample, from another row in the array.

This further demonstrates how to filter the audio source through thefirst filter to into a series of frequency-bands-with-noise, thensuppressing high energy bands to increase feedback in the series offrequency-bands-with-noise, then re-filtering the series offrequency-bands-with-noise through a second filter; and outputting theseries of frequency-bands-with-noise as audio output to produce musicalsound.

FIG. 4 a illustrates a block-diagram view showing an example of afrequency aperture filter in accordance with an embodiment; while FIG. 4b illustrates a block-diagram view showing another example of afrequency aperture filter in accordance with another embodiment. Thesefigures show how selected parameters to specify the desired frequencyslit spacing and the desired noise-to-frequency band ratio can be usedto filter and conform conforming the series offrequency-bands-with-noise to the parameters to produce the desiredfrequency slit spacing and the desired noise-to-frequency band ratio.

Before discussing frequency aperture filters, some analogous inspirationmay help understanding. White noise is a sound that covers the entirerange of audible frequencies, all of which possess similar intensity. Anapproximation to white noise is the static that appears between FM radiostations. Pink noise contains frequencies of the audible spectrum, butwith a decreasing intensity of roughly three decibels per octave. Thisdecrease approximates the audio spectrum composite of acoustic musicalinstruments or ensembles.

At least one embodiment of the invention was inspired by the way that aprism can separate white light into it's constituent spectrum offrequencies. White noise can be thought of as analogous to white light,which contains roughly equal intensities of all frequencies of visiblelight. A prism can separate white light into it's constituent spectrumof frequencies, the resultant frequencies based on the material,internal feedback interference and spectrum of incoming light.

Among other factors, frequency aperture cells (FACs) (See FIG. 3 andaccompanying text) do analogously with audio, based on their type,feedback properties, and the spectrum of incoming audio. Another aspectof an embodiment of the invention deals with the conversion of incomingpitched sounds into wide-band audio noise spectra, while at the sametime preserving the intelligibility, sibilance, or transient aspect ofthe original sound, then routing the sound through the array of FAC's.

In accordance with an embodiment, frequency aperture filters 300 (FAF)may be embodied as single or multiple digital filters of either the IIR.(infinite impulse Response) or FIR (Finite Impulse Response) type, orany combination thereof. One characteristic of the filters 300 is thatboth timbre and pitch are controlled by the filter parameters, and thatinput frequencies of adequate energies that line up with the multiplepass-bands of the filter 300 will be passed to the output of thecollective filter 300, albeit of potentially differing amplitude andphase.

In one example embodiment, an input impulse or other initializationenergy is preloaded into a multi-channel circular buffer 310. A bufferaddress control block calculates successive write addresses to preloadthe entire circular buffer with impulse transient energy whenever, forexample, a new note is depressed on the music keyboard.

The circular buffer arrangement allows for very efficient usage of theCPU and memory, which may reduce required amount of computer hardwareresources needed to perform real-time processing of the audio synthesis.In other embodiments, the efficient usage of computer resources allowsprocessing of the system and methods in a virtual computing environment,such as, a java virtual machine.

In accordance with an embodiment, Left and Right Stereo or mono audio isdemultiplexed into four channels, based on the combination type desiredfor the aperture spacing. This is the continuous live streaming audiothat follows the impulse transient loading.

After that, continuous, successive write addresses are generated by thebuffer address control for incoming combined input samples, as well asfor successive read addresses for outgoing samples into theInterpolation and Processing block 320 (See also FIG. 6).

In one example buffer address calculation, the read address isdetermined by the write address, by subtracting from it a base tuningreference value divided by the read pitch step size. The base tuningreference value is calculated from the FAF 300 filter type, via lookuptable or hard calculations, as different FAF 300 filter types change theoverall delay through the feedback path and are therefore pitchcompensated via this control. The same control is deployed to themulti-mode filter in the interpolate and processing block (See FIG. 6),as this variable filter contributes to the overall feedback delay whichcontributes to the perceived pitch through the FAF 300. The read stepsize is calculated from the slit_width 330 input. The pass bands of thefilter may be determined in part by the spacing of the read and writepointers, which represent the Infinite Impulse, or feedback portion ofan IIR filter design. The read address in this case may have both aninteger and fractional component, the later of which is used by theinterpolation and processing block 320.

Looking ahead to FIG. 6 illustrates a block-diagram view showing theinterpolate and process block of FIGS. 4 a and 4 b in accordance with anembodiment. In accordance with an embodiment, the Interpolate andProcess block 320 is used to lookup and calculate a value “in between”two successive buffer values at the audio sample rate. The interpolationmay be of any type, such as well known linear, spline, or sine(x)/xwindowed interpolation. By virtue of the quad interleave buffer, andcorresponding interleave coefficient and state variable data structures,four simultaneous calculations may be performed at once. In addition tointerpolation, the block processing includes filtering for high-pass,low-pass, or other tone shaping. The four interleave channels havediffering, filter types and coefficients, for musicality and enhancingstereo imaging. In addition, there may be multiple types ofinterpolation needed at once, one to resolve the audio sample rate rangevia up-sampling and down-sampling, and one to resolve the desiredslit_width.

Turning back to FIG. 5 illustrates a block-diagram view showing theselection and combination block of FIGS. 4 a and 4 b in accordance withan embodiment. The Selection and combination block 350 is comprised ofadaptive stability compensation filtering based on the desiredslit_width, slit height, and FAF Type. The audio frequency componentsfrom the Interpolate and Process block 320 are combined by applyingadaptive filtering as needed to attenuate the frequency bands of maximumamplitude, then mixing the harmonic-to-noise ratios together atdifferent amplitudes.

Turning ahead to FIG. 9 illustrates a block-diagram view showing thestability compensation filter of FIG. 5 in accordance with anembodiment. Shown is an example digital biquad filter, however, othertypes of stabilization techniques may be used. Stability compensationfiltering allows for maintaining stability and harmonic purity of arecursive IIR design at relatively higher values of slit_width and slitheight, which may be changing continuously in value. The stabilitycoefficients are adapted over time based on the changing values of keypitch, slit_height (harmonic/noise ratio), and slit width (frequencypartial spacing). For example, higher note pitch and wider slit_width(higher partial spacing) may generally require greater attenuation oflower frequency bands in order to maintain filter stability.

The stability compensation filter may calculate a co-efficient of thestability filter to prevent the system from passing of unity gain. A keytracker (also known as a key scaler) scales the incoming musical notekey according to linear or nonlinear functions which may be of simpletabular form. The stability compensation filter may use a key tracker inits calculations to determine the desired amount of noise-to-feedbackratio. The stability compensation fitter may use a key tracker todetermine the desired amount of frequency slit spacing (e.g. variationson slit_width).

Again on FIGS. 4 a and 4 b, after interpolation and processing 320, theaudio is multiplexed in the output mux and combination block 360. Theoutput multiplexing complements both-the input de-multiplexing and theselection and combination blocks to accumulate the desired output audiosignal and aperture spacing character.

FIG. 7 illustrates a block-diagram view showing one example of amulti-mode filter, which may be seen in FIGS. 1 and 2 in accordance withan embodiment. Multi-mode filters may be optionally used in frequencyaperture arrays. Examples of multi-mode filters include, high pass, lowpass, band pass, band restrict, and combinations. This demonstrates howmultimode filters the output of each filter in the sequence using amulti-mode-filter such as a lowpass filter, highpass filter, bandpassfilter, and bandreject filter.

FIG. 8 illustrates a block-diagram view showing various modulators inaccordance with an embodiment. The input audio signal itself can besubject to modulation by various methods including algorithmic means(random generators, low frequency oscillation (LFO) modulation, envelopemodulation, etc.), MIDI control means (MIDI Continuous Controllers, MIDINote messages, MIDI system messages, etc.); or physical controllerswhich output MIDI messages or analog voltage, as shown. Other modulationmethods may be possible as well.

FIG. 10 illustrates a block-diagram view showing how an audio inputsource into the FAA synthesizer can be modulated before entering the FAAfilters, and how the FAA filters themselves can be modulated inreal-time, in accordance with an embodiment. In particular, this showshow an audio input source into the FAA synthesizer may be modulatedbefore entering the FAA filters. It also shows how the FAA filtersthemselves can be modulated in real-time. In some embodiments the FAAsynthesis can be combined with other synthesis methods, in accordancewith various embodiments. In some embodiments, a console orkeyboard-like application may be employed, which can be used with thesystem as described herein.

FIG. 11 a illustrates a FFT spectral waveform graph view showing aslit_height of 100% in accordance with an embodiment; FIG. 11 billustrates a FFT spectral waveform graph view showing a slit_height of50% in accordance with an embodiment; FIG. 11 c illustrates a FFTspectral waveform graph view showing a slit_height of 0% in accordancewith an embodiment; FIG. 11 d illustrates a FFT spectral waveform graphview showing a slit_height of −50% in accordance with an embodiment; andFIG. 11 e illustrates a FFT spectral waveform graph view showing aslit_height of −100% in accordance with an embodiment. Taken together,FIGS. 11 a, 11 b, 11 d, and 11 e show how the spectral waveforms changeas a result of processing through a frequency aperture filter. Becauseslit_height is 0% in FIG. 11 c, it shows the unprocessed waveform (e.g.noise) that was use as input to the frequency aperture filter. Peaks canbe seen approximately every 200 db. The first peak varies by about oneoctave from 100% slit_height to −100% slit_height.

FIG. 12 illustrates a FFT spectral waveform graph view showing acomparison of brown noise and pink noise as audio input in accordancewith an embodiment. In this graph, it can be seen that the synthesizedbrown noise has less energy at higher frequencies (similar to the brownnoise input). By comparison, the pink noise has consistent energy levelsat higher frequencies (similar to the pink noise input).

FIG. 13 illustrates a FFT spectral waveform graph view showing a seriesof waveforms in a 1-series-by-2-parallel array, including waveforms foraudio input, waveforms for output from each FAC, and a final waveformfor audio output in accordance with an embodiment. In this series ofwaveforms, brown noise and white noise are shown as input. Afterprocessing through a frequency aperture cell, the resulting waveform isdisplayed. Finally, the combination of the two results is shown as theparallel additive composite.

FIG. 14 illustrates a FFT spectral waveform graph view showing a seriesof waveforms in a 2-series-by-1-parallel array, including a waveform foraudio input, waveforms for output from each FAC, and a final waveformfor audio output in accordance with an embodiment. In this series ofwaveforms, the input source is brown noise. After processing through thefirst FAF (of Type4_turbo), the resultant waveform is shown. Afterprocessing through a second FAF (of Type1_normal), the final waveform isshown. This exemplifies processing of audio signals through a series offrequency aperture filters.

FIG. 15 illustrates a FFT spectral waveform graph view showing a seriesof waveforms in a 1-series-by-2-parallel array, including identicalwaveforms for audio input, waveforms for output from each FAC, eachprocessed separately with a different FAF Type, and each showingdifferent final waveforms for audio output in accordance with anembodiment. These waveform graphs show the differences in filter types,given the same waveform input.

FIGS. 16, 17, 18, 19, and 20 illustrate a series of computer screenshotviews showing user controls to select parameters, such as slit_height,slit_width and other pre-sets, for use or initialization in the FACs inaccordance with an embodiment. This screenshots show how the user ofcomputer software can set the slit_width, slit_height, number and typeof frequency aperture cells, and other pre-sets to produce synthesizedaudio. The slit_width (i.e. the desired frequency slit spacing) and theslit_height (i.e. desired noise-to-frequency band ratio) may be selectedto produce a specific fibre or other musical quality. Then duringfilter, the series of frequency-bands-with-noise will be generated toconform to the selection.

Appendix A lists sets of parameters and other pre-sets to producevarious example timbres in accordance with an embodiment. Theseparameters and pre-sets may be available to the user of a computer ordisplayed on screens such as those shown in FIGS. 16, 17, 18, 19 and 20.

The above-described systems and methods can be used in accordance withvarious embodiments to provide a number of different applications,including but not limited to:

-   -   A system and method that can synthesize pitched, musical sounds        from non-pitched, broad-spectrum audio.    -   A system and method of combining and arranging frequency        aperture cells for extreme efficiency of processing and memory.    -   A system and method of transforming audio with discernible pitch        and timbre into broad-spectrum noise with no discernible pitch        and timbre.    -   A system and method for combining the above synthesis with other        synthesis methods to create hybrid synthesizers.    -   A system and method for modulating individual components of the        system using MIDI, algorithmic or physical controllers.    -   A system and method for using real-time, streamed audio as an        input audio source for the above synthesizer.    -   A system and method for vocalizing into the above synthesizer        while playing MIDI and having the vocalization re--pitched and        harmonized.    -   A system and method for inputting any musical audio source,        whether file-based or streamed, and re-pitching it and        re-harmonizing.    -   A system and method for vocalizing into the above synthesizer        while playing MIDI and having the synthesizer play a        recognizable musical instrument.

The present invention may be conveniently implemented using one or moreconventional general purpose or specialized digital computers ormicroprocessors programmed according to the teachings of the presentdisclosure. Appropriate software coding can readily be prepared byskilled programmers based on the teachings of the present disclosure, aswill be apparent to those skilled in the software art.

In some embodiments, the present invention includes a computer programproduct which is a storage medium (media) having instructions storedthereon/in which can be used to program a computer to perform any of theprocesses of the present invention. The storage medium can include, butis not limited to, any type of disk including floppy disks, opticaldiscs, DVD, CD-ROMs, microdrive, and magneto-optical disks, ROMs, RAMs,EPROMs, EEPROMs, DRAMs, VRAMs, flash memory devices, magnetic or opticalcards, nanosystems (including molecular memory ICs), or any type ofmedia or device suitable for storing instructions and/or data.

There are a total of 17 source code files incorporated by reference toan earlier application. Further, many other advantages of applicant'sinvention will be apparent to those skilled in the art from the computersoftware source code and included screen shots.

A portion of the disclosure of this patent document contains materialwhich is subject to copyright protection; i.e. Copyright 2010 James VanBuskirk (17 U.S.C. 401). The copyright owner has no objection to thefacsimile reproduction by anyone of the patent document or the patentdisclosure, as it appears in the Patent and Trademark Office patent fileor records, but otherwise reserves all copyright rights whatsoever.

The foregoing description of the present invention has been provided forthe purposes of illustration and description. It is not intended to beexhaustive or to limit the invention to the precise forms disclosed. Theembodiments were chosen and described in order to best explain theprinciples of the invention and its practical application, therebyenabling others skilled in the art to understand the invention forvarious embodiments and with various modifications that are suited tothe particular use contemplated. It is intended that the scope of theinvention be defined by the following claims and their equivalence.

1-25. (canceled)
 26. A method for synthesizing audio to produce amusical sound, comprising the steps of: receiving an audio source;setting parameters of at least two filters to specify a desiredfrequency slit spacing and the desired noise-to-frequency-band ratio,wherein the frequency slit spacing for each filter corresponds to one ofharmonic frequency bands, inharmonic frequency bands, or a combinationof harmonic and inharmonic frequency bands; filtering a signal based onthe audio source through a sequence of the at least two filters tofilter the audio source into a series of harmonic, inharmonic, or acombination of harmonic and inharmonic frequency-bands-with-noise;outputting an audio output to produce musical sound.
 27. The method ofclaim 1, wherein the audio source comprises pitched sounds, the methodfurther comprising: converting the pitched sounds of the audio sourceinto wide-band audio noise spectra, wherein intelligibility, sibilance,or transient aspects of the incoming audio is preserved during theconversion of the pitched sounds; and performing the filtering on theconverted audio source to produce the musical sound.
 28. The method ofclaim 1, wherein the audio source comprises sounds of a firstinstrument, the method further comprising: converting the pitched soundsof the audio source into wide-band audio noise spectra; and performingthe filtering on the converted audio source so that the audio outputsounds like a second instrument.
 29. The method of claim 1, wherein: thefiltering of the signal based on the audio source changes a first pitchin the audio source to a different second pitch in the output audio. 30.The method of claim 1, wherein the audio source comprises norm pitchedsounds, the method further comprising: modulating an output of one ofthe at least two filters, wherein modulation is triggered by an externalsource that provides pitch information; and outputting the audio outputto produce the musical sound based on the modulation.
 31. The method ofclaim 5, wherein: the modulation is triggered by a keyboard.
 32. Themethod of claim 5, wherein: the audio source comprises talking,whispering, or vocalization; and the output audio resembles a musicalinstrument, based on the modulation.
 33. The method of claim 1, whereinthe audio source comprises pitched sounds, the method furthercomprising: converting the pitched sounds of the audio source intowide-band audio noise spectra; performing the filtering on the convertedaudio source; modulating an output of one of the at least two filters,wherein modulation is triggered by an external source that providespitch information; and outputting the audio output to produce themusical sound based on the modulation.
 34. The method of claim 1,wherein: the frequency slit spacing for one of the at least two filterscorresponds to harmonic frequency bands.
 35. The method of claim 1,wherein: the frequency slit spacing for one of the at least two filterscorresponds to inharmonic frequency bands.
 36. A system for synthesizingaudio to produce a musical sound, comprising: an input interface forreceiving an audio source; two or more frequency aperture cells withconfigurable parameters corresponding to a desired frequency slitspacing and a desired noise-to-frequency-band ratio and configured tofilter a signal based on the audio source into a series of harmonic,inharmonic, or a combination of harmonic and inharmonicfrequency-bands-with-noise; and an output interface for outputting anaudio output to produce musical sound.
 37. The system of claim 11,wherein the audio source comprises pitched sounds, the system furthercomprising: a dispersion module for converting the pitched sounds of theaudio source into wide-band audio noise spectra, whereinintelligibility, sibilance, or transient aspects of the incoming audiois preserved during the conversion of the pitched sounds; and whereinthe frequency aperture cells filter the converted audio source toproduce the musical sound,
 38. The system of claim 11, wherein the audiosource comprises sounds of a first instrument, the system furthercomprising: a dispersion module for converting the pitched sounds of theaudio source into wide-band audio noise spectra; and wherein thefrequency aperture cells filter the converted audio source so that theaudio output sounds like a second instrument.
 39. The system of claim11, wherein: the filtering of the signal based on the audio sourcechanges a first pitch in the audio source to a different second pitch inthe output audio.
 40. The system of claim 11, wherein the audio sourcecomprises non pitched sounds, the system further comprising: an externalsource that provides pitch information and is configured to provide atrigger for modulating an output of one of the at least two filters; andwherein outputting the audio output to produce the musical sound isbased on the modulation.
 41. The system of claim 15, wherein: theexternal source that triggers modulation is a keyboard.
 42. The systemof claim 15, wherein: the audio source comprises talking, whispering, orvocalization; and the output audio resembles a musical instrument, basedon the modulation of the output of one of the at least two filters. 43.The system of claim 11, wherein the audio source comprises pitchedsounds, the system further comprising: a dispersion module forconverting the pitched sounds of the audio source into wide-band audionoise spectra, and wherein the frequency aperture cells filter theconverted audio source; an external source that provides pitchinformation and is configured to provide a trigger for modulating anoutput of one of the at least two filters; and wherein outputting theaudio output to produce the musical sound is based on the modulation.44. The system of claim 11, wherein: the frequency slit spacing for oneof the at least two filters corresponds to harmonic frequency bands. 45.The system of claim 11, wherein: the frequency slit spacing for one ofthe at least two filters corresponds to inharmonic frequency hands.